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Ending a bridged call with a touchtone
by Marc Orenberg 16 Dec '08

16 Dec '08

Hello. I'm trying to allow the A-leg of a bridged call to be able to press a touchtone to end the call. In my Python script, I set-up a DTMF callback function using setInputCallback, but it doesn't seem to have any effect during bridged calls. Is there another way to do this? Thanks.

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FS mod_fax
by Helmut Kuper 16 Dec '08

16 Dec '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I tried to compile mod_fax today with trunk from yesterday. A 'make' in FS trunk directory led to an error saying that libspandsp.la wasn't found in libs/spandsp/src. So I had to configure and compile (make) spandsp manually before compiling FS. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkk+Q18ACgkQ4tZeNddg3dw5GgCgmmuLCsAx+T7IzUPayqAXZDaa BO8AoLa5wUOBqaEG1pOG4Qow8r7J2NF7 =MOUJ -----END PGP SIGNATURE-----

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OpenZAP: "Received unhandled message 125"
by Peter P GMX 16 Dec '08

16 Dec '08

I have installed OpenZAP with a TE220 card and EuroISDN. When I bridge an outgoing call I get a "Received unhandled message 125" Any idea what that means? As far as I know there is no <external> result code 125 defined in the ISDN protocol. Best regards Peter See the following logs. 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 14 -------------------------------------------------------------------------------- [08 02 80 03 7d 08 04 82 e3 98 28 14 01 01] 2008-11-27 17:35:17 [DEBUG] Span:0 Q.931() Received message from Q.921 (ind 4, tei 0, size 18) MesType: 125, CRVFlag 1 (Terminator), CRV 3 (Dialect: 2) 2008-11-27 17:35:17 [DEBUG] Span:0 Q.931() Sending message to Layer4 (size: 114) 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[7d] Size:[114] CRV: 3 (0x3, CTX: Terminator) 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 70d230 (1:1) source isdn_data->channels_local_crv[0x3] 2008-11-27 17:35:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received unhandled message 125 (0x7d) 2008-11-27 17:35:17 [DEBUG] Span:0 Q.921() Sending frame >oz list API CALL [oz(list)] output: +OK span: 1 (span1) type: isdn chan_count: 31 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options none +OK span: 2 (span2) type: isdn chan_count: 31 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options none

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mod_ldap
by Helmut Kuper 16 Dec '08

16 Dec '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I updated to latest FS trunk today and tried to compile mod_ldap and got this errors: mkdir .libs ar cru .libs/libldap.a bind.o open.o result.o error.o compare.o search.o controls.o messages.o references.o extended.o cyrus.o modify.o add.o modrdn.o delete.o abandon.o sasl.o sbind.o kbind.o unbind.o cancel.o filter.o free.o sort.o passwd.o whoami.o getdn.o getentry.o getattr.o getvalues.o addentry.o request.o os-ip.o url.o sortctrl.o vlvctrl.o init.o options.o print.o string.o util-int.o schema.o charray.o tls.o os-local.o dnssrv.o utf-8.o utf-8-conv.o turn.o groupings.o txn.o ppolicy.o version.o ranlib .libs/libldap.a creating libldap.la (cd .libs && rm -f libldap.la && ln -s ../libldap.la libldap.la) cc -g -O2 -o apitest apitest.o ./.libs/libldap.a /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/liblber/.libs/liblber.a ../../libraries/liblber/.libs/liblber.a ../../libraries/liblutil/liblutil.a /usr/lib/libsasl2.a -ldl -lresolv ./.libs/libldap.a(os-ip.o): In function `ldap_pvt_is_socket_ready': /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/libldap/os-ip.c:194: warning: `sys_errlist' is deprecated; use `strerror' or `strerror_r' instead /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/libldap/os-ip.c:194: warning: `sys_nerr' is deprecated; use `strerror' or `strerror_r' instead /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_close': (.text+0x348): undefined reference to `db_strerror' /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open': (.text+0x3d0): undefined reference to `db_create' /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open': (.text+0x453): undefined reference to `db_strerror' /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_getdata': (.text+0x7e8): undefined reference to `db_strerror' /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata': (.text+0xa4e): undefined reference to `db_strerror' /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata': (.text+0xafa): undefined reference to `db_strerror' /usr/lib/libsasl2.a(digestmd5.o): In function `init_des': (.text+0x6c8): undefined reference to `DES_key_sched' /usr/lib/libsasl2.a(digestmd5.o): In function `init_des': When I change src/mod/directories/mod_ldap/Makefile line 3 to "LDAP=openldap-2.4.11" and putting openldap-2.4.11.tgz in libs/ manually it compiles without error, but with some warnings. Maybe the openldap-2.3.19 delivered by FS is too old? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklHp/MACgkQ4tZeNddg3dzjOgCfSExGU7PSWSdAxI4VCZIosrLJ 25kAoK0igc53iV1PhWkj5faAVOL78E8B =ny1E -----END PGP SIGNATURE-----

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Openzap ERROR can't dial
by fidibus83 16 Dec '08

16 Dec '08

Have nobody an idea?

2 1

Bind error
by Jonas Gauffin 16 Dec '08

16 Dec '08

I get a bind error for the RTP, can someone be kind and explain why? 2008-12-16 10:47:07 [DEBUG] sofia_glue.c:2388 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000]/[PCMA:8:8000] 2008-12-16 10:47:07 [DEBUG] sofia_glue.c:1596 sofia_glue_tech_set_codec() Set Codec sofia/internal/0236661201(a)sip-corporate2.tele2.se PCMA/8000 20 ms 160 samples 2008-12-16 10:47:07 [DEBUG] sofia_glue.c:2352 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2008-12-16 10:47:07 [DEBUG] sofia_glue.c:1814 sofia_glue_activate_rtp() AUDIO RTP [sofia/internal/0236661201(a)sip-corporate2.tele2.se] 192.168.1.112 port 17102 -> 130.244.5X.XX port 27354 codec: 8 ms: 20 2008-12-16 10:47:07 [DEBUG] switch_rtp.c:858 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2008-12-16 10:47:07 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media() Ring-Ready sofia/internal/0236661201(a)sip-corporate2.tele2.se! 2008-12-16 10:47:07 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media() Pre-Answer sofia/internal/0236661201(a)sip-corporate2.tele2.se! 2008-12-16 10:47:07 [DEBUG] switch_channel.c:1615 switch_channel_perform_pre_answer() sofia/external/0707992871(a)212.151.14X.XXX receive message [SWITCH_MESSAGE_INDICATE_PROGRESS] 2008-12-16 10:47:07 [INFO] mod_sofia.c:1253 sofia_receive_message() Asked to send early media by sofia/external/0707992871(a)212.151.14X.XXX 2008-12-16 10:47:07 [DEBUG] switch_channel.c:1585 switch_channel_perform_mark_pre_answered() Send signal sofia/external/0707992871(a)212.151.14X.XXX [BREAK] 2008-12-16 10:47:07 [DEBUG] sofia_glue.c:497 sofia_glue_ext_address_lookup() STUN Success [85.89.7X.XXX]:[33594] 2008-12-16 10:47:07 [DEBUG] sofia_glue.c:1814 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/0707992871(a)212.151.14X.XXX] 192.168.1.112 port 17108 -> 130.244.1X.XXX port 17148 codec: 8 ms: 20 2008-12-16 10:47:07 [DEBUG] switch_rtp.c:858 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2008-12-16 10:47:07 [ERR] sofia_glue.c:2045 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-16 10:47:07 [NOTICE] sofia_glue.c:2046 sofia_glue_activate_rtp() Hangup sofia/external/0707992871(a)212.151.14X.XXX [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2008-12-16 10:47:07 [DEBUG] switch_channel.c:1478 switch_channel_perform_hangup() Send signal sofia/external/0707992871(a)212.151.14X.XXX [KILL] 2008-12-16 10:47:07 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/0707992871(a)212.151.14X.XXX [BREAK] 2008-12-16 10:47:07 [INFO] mod_sofia.c:1294 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1229387233 1229387234 IN IP4 85.89.7X.XXX s=FreeSWITCH c=IN IP4 85.89.7X.XXX t=0 0 m=audio 33594 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2008-12-16 10:47:07 [NOTICE] mod_sofia.c:1297 sofia_receive_message() Ring-Ready sofia/external/0707992871(a)212.151.14X.XXX! 2008-12-16 10:47:07 [NOTICE] mod_sofia.c:1297 sofia_receive_message() Pre-Answer sofia/external/0707992871(a)212.151.14X.XXX! 2008-12-16 10:47:07 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/external/0707992871(a)212.151.14X.XXX [BREAK] 2008-12-16 10:47:07 [DEBUG] switch_ivr_originate.c:1625 switch_ivr_originate() Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2008-12-16 10:47:07 [NOTICE] switch_ivr_originate.c:1666 switch_ivr_originate() Hangup sofia/internal/ 0236661201(a)sip-corporate2.tele2.se [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2008-12-16 10:47:07 [DEBUG] switch_channel.c:1478 switch_channel_perform_hangup() Send signal sofia/internal/ 0236661201(a)sip-corporate2.tele2.se [KILL] 2008-12-16 10:47:07 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/internal/ 0236661201(a)sip-corporate2.tele2.se [BREAK] 2008-12-16 10:47:07 [INFO] mod_dptools.c:1869 audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL

2 1

Bridging through gateway
by Jonas Gauffin 16 Dec '08

16 Dec '08

I'm trying to bridge using a non-registered gateway. And I get MANDATORY_IE_MISSING back. Why is that? 2008-12-15 11:06:49 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/u1000044@192.168.1.112:5070Execute bridge(sofia/default/ 0236661201(a)sip-corporate2.tele2.se) [.....] 2008-12-15 11:06:49 [DEBUG] sofia.c:2511 sofia_handle_sip_i_state() Channel sofia/internal/0236661201(a)sip-corporate2.tele2.se entering state [calling] 2008-12-15 11:06:49 [ERR] sofia_reg.c:1312 sofia_reg_handle_sip_r_challenge() No Matching gateway found 2008-12-15 11:06:49 [NOTICE] sofia_reg.c:1333 sofia_reg_handle_sip_r_challenge() Hangup sofia/internal/ 0236661201(a)sip-corporate2.tele2.se [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] 2008-12-15 11:06:49 [DEBUG] switch_channel.c:1478 switch_channel_perform_hangup() Send signal sofia/internal/ 0236661201(a)sip-corporate2.tele2.se [KILL]

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running custom script with bind_meta_app
by Stephen Crosby 16 Dec '08

16 Dec '08

I'm Stephen Crosby, and I've just started working with freeswitch. It's been great so far. I want to run a custom script inside a conference when a DTMF sequence is entered. I found bind_meta_app and thought it would be perfect, but I can't seem to get it to work. When I dial-in and press *8, I get no debugging output at all. When I press another sequence like *9 for instance, I get: [WARNING] switch_ivr_async.c:1429 meta_on_dtmf() sofia/external/5593495805(a)sip.gafachi.com Ignoring meta digit '9' not mapped. The script I wrote has been tested with "jsrun script.js" from the command line and it does work. I've got the debugging level all the way up and there's just not much for me to go on. Any help would be greatly appreciated. <extension name="gafachi-DID1"> <condition field="destination_number" expression="^xxxxxxxxxxx$"> <action application="answer"/> <action application="bind_meta_app" data="8 ab s javascript::$${base_dir}/scripts/script.js"/> <action application="conference" data="3000@default"/> </condition> </extension> --Stephen

3 3

Building a web config/billing gui
by ccav 16 Dec '08

16 Dec '08

For anyone interested, I'm in the process of building a web based config/billing gui. I could use some help though. Anyone who has php experience and knows the xml_curl interface pretty well and some spare time to do some development would be a useful partner. Also, if there's a resource online that defines all the params/variables and subobjects for each of the fs object types, like directories, dialplans etc, it would help me fill my parm database a lot faster. For anyone interested in monitoring the development, it's at www.sparkz.tv/smfs login is demo:demo. Feedback on features and development direction is invited. I want to get this done quickly. I think FS is awesome but no config/billing interface is going to stand in the way of it's adoption so I'm burning the midnight oil to get this done. -- View this message in context: http://www.nabble.com/Building-a-web-config-billing-gui-tp21027515p21027515… Sent from the Freeswitch-users mailing list archive at Nabble.com.

3 2

(no subject)
by martin joseph 15 Dec '08

15 Dec '08

I am wondering if anyone running OSX 10.4.11 is able to successfully build FS Trunk? If so, what versions does build.sh report? Please? I get the following and have issues making it... Updated to revision 10753. mail:/usr/src/freeswitch/trunk root# ./bootstrap.sh bootstrap: checking installation... bootstrap: autoconf version 2.61 (ok) bootstrap: automake version 1.10 (ok) bootstrap: libtool version 1.5.24 (ok) Thanks for the info, Marty

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